Updates: 2736 May 2017 Category: Informational
ISSN: 2070-1721
How to Write an RTP Payload Format Abstract
This document contains information on how best to write an RTP payload format specification. It provides reading tips, design
practices, and practical tips on how to produce an RTP payload format specification quickly and with good results. A template is also included with instructions.
Status of This Memo
This document is not an Internet Standards Track specification; it is published for informational purposes.
This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Not all documents approved by the IESG are a candidate for any level of Internet Standard; see Section 2 of RFC 7841.
Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc8088.
Copyright Notice
Copyright (c) 2017 IETF Trust and the persons identified as the document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust’s Legal Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents
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Table of Contents
1. Introduction ...4
1.1. Structure ...4
2. Terminology ...5
2.1. Definitions ...5
2.2. Abbreviations ...5
2.3. Use of Normative Requirements Language ...6
3. Preparations ...6
3.1. Read and Understand the Media Coding Specification ...6
3.2. Recommended Reading ...7
3.2.1. IETF Process and Publication ...7
3.2.2. RTP ...9
3.3. Important RTP Details ...13
3.3.1. The RTP Session ...13
3.3.2. RTP Header ...14
3.3.3. RTP Multiplexing ...16
3.3.4. RTP Synchronization ...16
3.4. Signaling Aspects ...18
3.4.1. Media Types ...19
3.4.2. Mapping to SDP ...20
3.5. Transport Characteristics ...23
3.5.1. Path MTU ...23
3.5.2. Different Queuing Algorithms ...23
3.5.3. Quality of Service ...24
4. Standardization Process for an RTP Payload Format ...24
4.1. IETF ...25
4.1.1. Steps from Idea to Publication ...25
4.1.2. WG Meetings ...27
4.1.3. Draft Naming ...27
4.1.4. Writing Style ...28
4.1.5. How to Speed Up the Process ...29
4.2. Other Standards Bodies ...29
4.3. Proprietary and Vendor Specific ...30
4.4. Joint Development of Media Coding Specification and RTP Payload Format ...31
5. Designing Payload Formats ...31
5.1. Features of RTP Payload Formats ...32
5.1.1. Aggregation ...32
5.1.2. Fragmentation ...33
5.1.3. Interleaving and Transmission Rescheduling ...33
5.1.4. Media Back Channels ...34
5.1.5. Media Scalability ...34
5.1.6. High Packet Rates ...37
5.2. Selecting Timestamp Definition ...37
6. Noteworthy Aspects in Payload Format Design ...39
6.1. Audio Payloads ...39
6.2. Video ...40
6.3. Text ...41
6.4. Application ...41
7. Important Specification Sections ...42
7.1. Media Format Description ...42
7.2. Security Considerations ...43
7.3. Congestion Control ...44
7.4. IANA Considerations ...45
8. Authoring Tools ...45
8.1. Editing Tools ...46
8.2. Verification Tools ...46
9. Security Considerations ...47
10. Informative References ...47
Appendix A. RTP Payload Format Template ...58
A.1. Title ...58
A.2. Front-Page Boilerplate ...58
A.3. Abstract ...58
A.4. Table of Contents ...58
A.5. Introduction ...59
A.6. Conventions, Definitions, and Abbreviations ...59
A.7. Media Format Description ...59
A.8. Payload Format ...59
A.8.1. RTP Header Usage ...59
A.8.2. Payload Header ...59
A.8.3. Payload Data ...60
A.9. Payload Examples ...60
A.10. Congestion Control Considerations ...60
A.11. Payload Format Parameters ...60
A.11.1. Media Type Definition ...60
A.11.2. Mapping to SDP ...62
A.12. IANA Considerations ...63
A.13. Security Considerations ...63
A.14. RFC Editor Considerations ...64
A.15. References ...64
A.15.1. Normative References ...64
A.15.2. Informative References ...64
A.16. Authors’ Addresses ...64
Acknowledgements ...64
Contributors ...65
Author’s Address ...65
1. Introduction
RTP [RFC3550] payload formats define how a specific real-time data format is structured in the payload of an RTP packet. A real-time data format without a payload format specification cannot be
transported using RTP. This creates an interest in many individuals/
organizations with media encoders or other types of real-time data to define RTP payload formats. However, the specification of a well- designed RTP payload format is nontrivial and requires knowledge of both RTP and the real-time data format.
This document is intended to help any author of an RTP payload format specification make important design decisions, consider important features of RTP and RTP security, etc. The document is also intended to be a good starting point for any person with little experience in the IETF and/or RTP to learn the necessary steps.
This document extends and updates the information that is available in "Guidelines for Writers of RTP Payload Format Specifications"
[RFC2736]. Since that RFC was written, further experience has been gained on the design and specification of RTP payload formats.
Several new RTP profiles and robustness tools have been defined, and these need to be considered.
This document also discusses the possible venues for defining an RTP payload format: the IETF, other standards bodies, and proprietary ones.
Note, this document does discuss IETF, IANA, and RFC Editor processes and rules as they were when this document was published. This to make clear how the work to specify an RTP payload formats depends, uses, and interacts with these rules and processes. However, these rules and processes are subject to change and the formal rule and process specifications always takes precedence over what is written here.
1.1. Structure
This document has several different parts discussing different aspects of the creation of an RTP payload format specification.
Section 3 discusses the preparations the author(s) should make before starting to write a specification. Section 4 discusses the different processes used when specifying and completing a payload format, with focus on working inside the IETF. Section 5 discusses the design of payload formats themselves in detail. Section 6 discusses current design trends and provides good examples of practices that should be followed when applicable. Following that, Section 7 provides a discussion on important sections in the RTP payload format
specification itself such as Security Considerations and IANA Considerations. This document ends with an appendix containing a template that can be used when writing RTP payload formats
specifications.
2. Terminology 2.1. Definitions
RTP Stream: A sequence of RTP packets that together carry part or all of the content of a specific media (audio, video, text, or data whose form and meaning are defined by a specific real-time application) from a specific sender source within a given RTP session.
RTP Session: An association among a set of participants
communicating with RTP. The distinguishing feature of an RTP session is that each session maintains a full, separate space of synchronization source (SSRC) identifiers. See also
Section 3.3.1.
RTP Payload Format: The RTP payload format specifies how units of a specific encoded media are put into the RTP packet payloads and how the fields of the RTP packet header are used, thus enabling the format to be used in RTP applications.
A Taxonomy of Semantics and Mechanisms for Real-Time Transport Protocol (RTP) Sources [RFC7656] defines many useful terms.
2.2. Abbreviations
ABNF: Augmented Backus-Naur Form [RFC5234]
ADU: Application Data Unit ALF: Application Level Framing ASM: Any-Source Multicast BCP: Best Current Practice I-D: Internet-Draft
IESG: Internet Engineering Steering Group MTU: Maximum Transmission Unit
QoS: Quality of Service RFC: Request For Comments
RTP: Real-time Transport Protocol RTCP: RTP Control Protocol
RTT: Round-Trip Time
SSM: Source-Specific Multicast
2.3. Use of Normative Requirements Language
As this document is both Informational and instructional rather than a specification, this document does not use any RFC 2119 language and the use of "may", "should", "recommended", and "must" carries no special connotation.
3. Preparations
RTP is a complex real-time media delivery framework, and it has a lot of details that need to be considered when writing an RTP payload format. It is also important to have a good understanding of the media codec / format so that all of its important features and properties are considered. Only when one has sufficient
understanding of both parts can one produce an RTP payload format of high quality. On top of this, one needs to understand the process within the IETF and especially the Working Group responsible for standardizing payload formats (currently the PAYLOAD WG) to go
quickly from the initial idea stage to a finished RFC. This and the next sections help an author prepare himself in those regards.
3.1. Read and Understand the Media Coding Specification
It may be obvious, but it is necessary for an author of an RTP payload specification to have a solid understanding of the media to be transported. Important are not only the specifically spelled out transport aspects (if any) in the media coding specification, but also core concepts of the underlying technology. For example, an RTP payload format for video coded with inter-picture prediction will perform poorly if the payload designer does not take the use of inter-picture prediction into account. On the other hand, some (mostly older) media codecs offer error-resilience tools against bit errors, which, when misapplied over RTP, in almost all cases would only introduce overhead with no measurable return.
3.2. Recommended Reading
The following subsections list a number of documents. Not all need to be read in full detail. However, an author basically needs to be aware of everything listed below.
3.2.1. IETF Process and Publication
Newcomers to the IETF are strongly recommended to read the "Tao of the IETF" [TAO] that goes through most things that one needs to know about the IETF: the history, organizational structure, how the WGs and meetings work, etc.
It is very important to note and understand the IETF Intellectual Property Rights (IPR) policy that requires early disclosures based on personal knowledge from anyone contributing in IETF. The IETF
policies associated with IPR are documented in BCP 78 [BCP78]
(related to copyright, including software copyright, for example, code) and BCP 79 [BCP79] (related to patent rights). These rules may be different from other standardization organizations. For example, a person that has a patent or a patent application that he or she reasonably and personally believes to cover a mechanism that gets added to the Internet-Draft they are contributing to (e.g., by submitting the draft, posting comments or suggestions on a mailing list, or speaking at a meeting) will need to make a timely IPR disclosure. Read the above documents for the authoritative rules.
Failure to follow the IPR rules can have dire implications for the specification and the author(s) as discussed in [RFC6701].
Note: These IPR rules apply on what is specified in the RTP
payload format Internet-Draft (and later RFC); an IPR that relates to a codec specification from an external body does not require IETF IPR disclosure. Informative text explaining the nature of the codec would not normally require an IETF IPR declaration.
Appropriate IPR declarations for the codec itself would normally be found in files of the external body defining the codec, in accordance with that external body’s own IPR rules.
The main part of the IETF process is formally defined in BCP 9 [BCP9]. BCP 25 [BCP25] describes the WG process, the relation between the IESG and the WG, and the responsibilities of WG Chairs and participants.
It is important to note that the RFC Series contains documents of several different publication streams as defined by The RFC Series and RFC Editor [RFC4844]. The most important stream for RTP payload
different categories: Standards Track, Informational, Experimental, Best Current Practice, and Historic. "Standards Track" contains two maturity levels: Proposed Standard and Internet Standard [RFC6410].
A Standards Track document must start as a Proposed Standard; after successful deployment and operational experience with at least two implementations, it can be moved to an Internet Standard. The Independent Submission Stream could appear to be of interest as it provides a way of publishing documents of certain categories such as Experimental and Informational with a different review process.
However, as long as IETF has a WG that is chartered to work on RTP payload formats, this stream should not be used.
As the content of a given RFC is not allowed to change once
published, the only way to modify an RFC is to write and publish a new one that either updates or replaces the old one. Therefore, whether reading or referencing an RFC, it is important to consider both the Category field in the document header and to check if the RFC is the latest on the subject and still valid. One way of
checking the current status of an RFC is to use the RFC Editor’s RFC search page (https://www.rfc-editor.org/search), which displays the current status and which if any RFC has updated or obsoleted it. The RFC Editor search engine will also indicate if there exist any errata reports for the RFC. Any verified errata report contains issues of significant importance with the RFC; thus, they should be known prior to an update and replacement publication.
Before starting to write a draft, one should also read the Internet- Draft writing guidelines (http://www.ietf.org/ietf/1id-
guidelines.txt), the I-D checklist (http://www.ietf.org/ID-
Checklist.html), and the RFC Style Guide [RFC7322]. Another document that can be useful is "Guide for Internet Standards Writers"
[RFC2360].
There are also a number of documents to consider in the process of writing drafts intended to become RFCs. These are important when writing certain types of text.
RFC 2606: When writing examples using DNS names in Internet-Drafts, those names shall be chosen from the example.com, example.net, and example.org domains.
RFC 3849: Defines the range of IPv6 unicast addresses (2001:DB8::/32) that should be used in any examples.
RFC 5737: Defines the ranges of IPv4 unicast addresses reserved for documentation and examples: 192.0.2.0/24, 198.51.100.0/24, and 203.0.113.0/24.
RFC 5234: Augmented Backus-Naur Form (ABNF) is often used when writing text field specifications. Not commonly used in RTP payload formats, but may be useful when defining media type parameters of some complexity.
3.2.2. RTP
The recommended reading for RTP consists of several different parts:
design guidelines, the RTP protocol, profiles, robustness tools, and media-specific recommendations.
Any author of RTP payload formats should start by reading "Guidelines for Writers of RTP Payload Format Specifications" [RFC2736], which contains an introduction to the Application Level Framing (ALF) principle, the channel characteristics of IP channels, and design guidelines for RTP payload formats. The goal of ALF is to be able to transmit Application Data Units (ADUs) that are independently usable by the receiver in individual RTP packets, thus minimizing
dependencies between RTP packets and the effects of packet loss.
Then, it is advisable to learn more about the RTP protocol, by studying the RTP specification "RTP: A Transport Protocol for Real- Time Applications" [RFC3550] and the existing profiles. As a
complement to the Standards Track documents, there exists a book totally dedicated to RTP [CSP-RTP]. There exist several profiles for RTP today, but all are based on "RTP Profile for Audio and Video Conferences with Minimal Control" [RFC3551] (abbreviated as RTP/AVP).
The other profiles that one should know about are "The Secure Real- time Transport Protocol (SRTP)" (RTP/SAVP) [RFC3711], "Extended RTP Profile for RTCP-based Feedback (RTP/AVPF)" [RFC4585], and "Extended Secure Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)" [RFC5124]. It is important to understand RTP and the RTP/AVP profile in detail. For the other profiles, it is sufficient to have an understanding of what functionality they provide and the limitations they create.
A number of robustness tools have been developed for RTP. The tools are for different use cases and real-time requirements.
RFC 2198: "RTP Payload for Redundant Audio Data" [RFC2198] provides functionalities to transmit redundant copies of audio or text payloads. These redundant copies are sent together with a primary format in the same RTP payload. This format relies on the RTP timestamp to determine where data belongs in a sequence;
therefore, it is usually most suitable to be used with audio.
However, the RTP Payload format for T.140 [RFC4103] text format
original sequence number. This makes it unusable for most video formats. This format is also only suitable for media formats that produce relatively small RTP payloads.
RFC 6354: The "Forward-Shifted RTP Redundancy Payload Support"
[RFC6354] is a variant of RFC 2198 that allows the redundant data to be transmitted prior to the original.
RFC 5109: The "RTP Payload Format for Generic Forward Error Correction" [RFC5109] provides an XOR-based Forward Error
Correction (FEC) of the whole or parts of a number of RTP packets.
This specification replaced the previous specification for XOR- based FEC [RFC2733]. These FEC packets are sent in a separate stream or as a redundant encoding using RFC 2198. This FEC scheme has certain restrictions in the number of packets it can protect.
It is suitable for applications with low-to-medium delay tolerance with a limited amount of RTP packets.
RFC 6015: "RTP Payload Format for 1-D Interleaved Parity Forward Error Correction (FEC)" [RFC6015] provides a variant of the XOR- based Generic protection defined in [RFC2733]. The main
difference is to use interleaving scheme on which packets gets included as source packets for a particular protection packet.
The interleaving is defined by using every L packets as source data and then producing protection data over D number of packets.
Thus, each block of D x L source packets will result in L number of Repair packets, each capable of repairing one loss. The goal is to provide better burst-error robustness when the packet rate is higher.
FEC Framework: "Forward Error Correction (FEC) Framework" [RFC6363]
defines how to use FEC protection for arbitrary packet flows.
This framework can be applied for RTP/RTCP packet flows, including using RTP for transmission of repair symbols, an example is in "RTP Payload Format for Raptor Forward Error Correction (FEC)"
[RFC6682].
RTP Retransmission: The RTP retransmission scheme [RFC4588] is used for semi-reliability of the most important RTP packets in a RTP stream. The level of reliability between semi- and in-practice full reliability depends on the targeted properties and situation where parameters such as round-trip time (RTT) allowed additional overhead and allowable delay. It often requires the application to be quite delay tolerant as a minimum of one round-trip time plus processing delay is required to perform a retransmission.
Thus, it is mostly suitable for streaming applications but may also be usable in certain other cases when operating in networks with short round-trip times.
RTP over TCP: RFC 4571 [RFC4571] defines how one sends RTP and RTCP packets over connection-oriented transports like TCP. If one uses TCP, one gets reliability for all packets but loses some of the real-time behavior that RTP was designed to provide. Issues with TCP transport of real-time media include head-of-line blocking and wasting resources on retransmission of data that is already late.
TCP is also limited to point-to-point connections, which further restricts its applicability.
There have been both discussion and design of RTP payload formats, e.g., Adaptive Multi-Rate (AMR) and AMR Wideband (AMR-WB) [RFC4867], supporting the unequal error detection provided by UDP-Lite
[RFC3828]. The idea is that by not having a checksum over part of the RTP payload one can allow bit errors from the lower layers. By allowing bit errors one can increase the efficiency of some link layers and also avoid unnecessary discarding of data when the payload and media codec can get at least some benefit from the data. The main issue is that one has no idea of the level of bit errors present in the unprotected part of the payload. This makes it hard or
impossible to determine whether or not one can design something usable. Payload format designers are not recommended to consider features for unequal error detection using UDP-Lite unless very clear requirements exist.
There also exist some management and monitoring extensions.
RFC 2959: The RTP protocol Management Information Database (MIB) [RFC2959] that is used with SNMP [RFC3410] to configure and retrieve information about RTP sessions.
RFC 3611: The RTCP Extended Reports (RTCP XR) [RFC3611] consists of a framework for reports sent within RTCP. It can easily be
extended by defining new report formats, which has and is
occurring. The XRBLOCK WG in the IETF is chartered (at the time of writing) with defining new report formats. The list of
specified formats is available in IANA’s RTCP XR Block Type registry (http://www.iana.org/assignments/rtcp-xr-block-types/).
The report formats that are defined in RFC 3611 provide report information on packet loss, packet duplication, packet reception times, RTCP statistics summary, and VoIP Quality. [RFC3611] also defines a mechanism that allows receivers to calculate the RTT to other session participants when used.
RMONMIB: The Remote Network Monitoring WG has defined a mechanism [RFC3577] based on usage of the MIB that can be an alternative to RTCP XR.
A number of transport optimizations have also been developed for use in certain environments. They are all intended to be transparent and do not require special consideration by the RTP payload format
writer. Thus, they are primarily listed here for informational reasons.
RFC 2508: "Compressing IP/UDP/RTP Headers for Low-Speed Serial Links" (CRTP) [RFC2508] is the first IETF-developed RTP header compression mechanism. It provides quite good compression;
however, it has clear performance problems when subject to packet loss or reordering between compressor and decompressor.
RFCs 3095 and 5795: These are the base specifications of the robust header compression (ROHC) protocol version 1 [RFC3095] and version 2 [RFC5795]. This solution was created as a result of CRTP’s lack of performance when compressed packets are subject to loss.
RFC 3545: Enhanced compressed RTP (E-CRTP) [RFC3545] was developed to provide extensions to CRTP that allow for better performance over links with long RTTs, packet loss, and/or reordering.
RFC 4170: "Tunneling Multiplexed Compressed RTP (TCRTP)" [RFC4170]
is a solution that allows header compression within a tunnel carrying multiple multiplexed RTP flows. This is primarily used in voice trunking.
There exist a couple of different security mechanisms that may be used with RTP. By definition, generic mechanisms are transparent for the RTP payload format and do not need special consideration by the format designer. The main reason that different solutions exist is that different applications have different requirements; thus, different solutions have been developed. For more discussion on this, please see "Options for Securing RTP Sessions" [RFC7201] and "Securing the RTP Framework: Why RTP Does Not Mandate a Single Media Security Solution" [RFC7202]. The main properties for an RTP
security mechanism are to provide confidentiality for the RTP
payload, integrity protection to detect manipulation of payload and headers, and source authentication. Not all mechanisms provide all of these features, a point that will need to be considered when a specific mechanisms is chosen.
The profile for Secure RTP - SRTP (RTP/SAVP) [RFC3711] and the derived profile (RTP/SAVPF [RFC5124]) are a solution that enables confidentiality, integrity protection, replay protection, and partial source authentication. It is the solution most commonly used with RTP at the time of writing this document. There exist several key- management solutions for SRTP, as well other choices, affecting the
security properties. For a more in-depth review of the options and solutions other than SRTP consult "Options for Securing RTP Sessions"
[RFC7201].
3.3. Important RTP Details
This section reviews a number of RTP features and concepts that are available in RTP, independent of the payload format. The RTP payload format can make use of these when appropriate, and even affect the behavior (RTP timestamp and marker bit), but it is important to note that not all features and concepts are relevant to every payload format. This section does not remove the necessity to read up on RTP. However, it does point out a few important details to remember when designing a payload format.
3.3.1. The RTP Session
The definition of the RTP session from RFC 3550 is:
An association among a set of participants communicating with RTP.
A participant may be involved in multiple RTP sessions at the same time. In a multimedia session, each medium is typically carried in a separate RTP session with its own RTCP packets unless the encoding itself multiplexes multiple media into a single data stream. A participant distinguishes multiple RTP sessions by reception of different sessions using different pairs of destination transport addresses, where a pair of transport
addresses comprises one network address plus a pair of ports for RTP and RTCP. All participants in an RTP session may share a common destination transport address pair, as in the case of IP multicast, or the pairs may be different for each participant, as in the case of individual unicast network addresses and port pairs. In the unicast case, a participant may receive from all other participants in the session using the same pair of ports, or may use a distinct pair of ports for each.
The distinguishing feature of an RTP session is that each session maintains a full, separate space of SSRC identifiers (defined next). The set of participants included in one RTP session
consists of those that can receive an SSRC identifier transmitted by any one of the participants either in RTP as the SSRC or a CSRC (also defined below) or in RTCP. For example, consider a three- party conference implemented using unicast UDP with each
participant receiving from the other two on separate port pairs.
If each participant sends RTCP feedback about data received from one other participant only back to that participant, then the
reception of one other participant to both of the other
participants, then the conference is composed of one multi-party RTP session. The latter case simulates the behavior that would occur with IP multicast communication among the three
participants.
The RTP framework allows the variations defined here, but a particular control protocol or application design will usually impose constraints on these variations.
3.3.2. RTP Header
The RTP header contains a number of fields. Two fields always require additional specification by the RTP payload format, namely the RTP timestamp and the marker bit. Certain RTP payload formats also use the RTP sequence number to realize certain functionalities, primarily related to the order of their application data units. The payload type is used to indicate the used payload format. The SSRC is used to distinguish RTP packets from multiple senders and media sources identifying the RTP stream. Finally, [RFC5285] specifies how to transport payload format independent metadata relating to the RTP packet or stream.
Marker Bit: A single bit normally used to provide important
indications. In audio, it is normally used to indicate the start of a talk burst. This enables jitter buffer adaptation prior to the beginning of the burst with minimal audio quality impact. In video, the marker bit is normally used to indicate the last packet part of a frame. This enables a decoder to finish decoding the picture, where it otherwise may need to wait for the next packet to explicitly know that the frame is finished.
Timestamp: The RTP timestamp indicates the time instance the media sample belongs to. For discrete media like video, it normally indicates when the media (frame) was sampled. For continuous media, it normally indicates the first time instance the media present in the payload represents. For audio, this is the
sampling time of the first sample. All RTP payload formats must specify the meaning of the timestamp value and the clock rates allowed. Selecting a timestamp rate is an active design choice and is further discussed in Section 5.2.
Discontinuous Transmission (DTX) that is common among speech
codecs, typically results in gaps or jumps in the timestamp values due to that there is no media payload to transmit and the next used timestamp value represent the actual sampling time of the data transmitted.
Sequence Number: The sequence number is monotonically increasing and is set as the packet is sent. This property is used in many
payload formats to recover the order of everything from the whole stream down to fragments of application data units (ADUs) and the order they need to be decoded. Discontinuous transmissions do not result in gaps in the sequence number, as it is monotonically increasing for each sent RTP packet.
Payload Type: The payload type is used to indicate, on a per-packet basis, which format is used. The binding between a payload type number and a payload format and its configuration are dynamically bound and RTP session specific. The configuration information can be bound to a payload type value by out-of-band signaling
(Section 3.4). An example of this would be video decoder configuration information. Commonly, the same payload type is used for a media stream for the whole duration of a session.
However, in some cases it may be necessary to change the payload format or its configuration during the session.
SSRC: The synchronization source (SSRC) identifier is normally not used by a payload format other than to identify the RTP timestamp and sequence number space a packet belongs to, allowing
simultaneously reception of multiple media sources. However, some of the RTP mechanisms for improving resilience to packet loss uses multiple SSRCs to separate original data and repair or redundant data, as well as multi-stream transmission of scalable codecs.
Header Extensions: RTP payload formats often need to include metadata relating to the payload data being transported. Such metadata is sent as a payload header, at the start of the payload section of the RTP packet. The RTP packet also includes space for a header extension [RFC5285]; this can be used to transport
payload format independent metadata, for example, an SMPTE time code for the packet [RFC5484]. The RTP header extensions are not intended to carry headers that relate to a particular payload format, and must not contain information needed in order to decode the payload.
The remaining fields do not commonly influence the RTP payload
format. The padding bit is worth clarifying as it indicates that one or more bytes are appended after the RTP payload. This padding must be removed by a receiver before payload format processing can occur.
Thus, it is completely separate from any padding that may occur within the payload format itself.
3.3.3. RTP Multiplexing
RTP has three multiplexing points that are used for different
purposes. A proper understanding of this is important to correctly use them.
The first one is separation of RTP streams of different types or usages, which is accomplished using different RTP sessions. So, for example, in the common multimedia session with audio and video, RTP commonly multiplexes audio and video in different RTP sessions. To achieve this separation, transport-level functionalities are used, normally UDP port numbers. Different RTP sessions can also be used to realize layered scalability as it allows a receiver to select one or more layers for multicast RTP sessions simply by joining the
multicast groups over which the desired layers are transported. This separation also allows different Quality of Service (QoS) to be
applied to different media types. Use of multiple transport flows has potential issues due to NAT and firewall traversal. The choices how one applies RTP sessions as well as transport flows can affect the transport properties an RTP media stream experiences.
The next multiplexing point is separation of different RTP streams within an RTP session. Here, RTP uses the SSRC to identify
individual sources of RTP streams. An example of individual media sources would be the capture of different microphones that are carried in an RTP session for audio, independently of whether they are connected to the same host or different hosts. There also exist cases where a single media source, is transmitted using multiple RTP streams. For each SSRC, a unique RTP sequence number and timestamp space is used.
The third multiplexing point is the RTP header payload type field.
The payload type identifies what format the content in the RTP
payload has. This includes different payload format configurations, different codecs, and also usage of robustness mechanisms like the one described in RFC 2198 [RFC2198].
3.3.4. RTP Synchronization
There are several types of synchronization, and we will here describe how RTP handles the different types:
Intra media: The synchronization within a media stream from a synchronization source (SSRC) is accomplished using the RTP
timestamp field. Each RTP packet carries the RTP timestamp, which specifies the position in time of the media payload contained in this packet relative to the content of other RTP packets in the same RTP stream (i.e., a given SSRC). This is especially useful
in cases of discontinuous transmissions. Discontinuities can be caused by network conditions; when extensive losses occur the RTP timestamp tells the receiver how much later than previously
received media the present media should be played out.
Inter-media: Applications commonly have a desire to use several media sources, possibly of different media types, at the same time. Thus, there exists a need to synchronize different media from the same endpoint. This puts two requirements on RTP: the possibility to determine which media are from the same endpoint and if they should be synchronized with each other and the functionality to facilitate the synchronization itself.
The first step in inter-media synchronization is to determine which SSRCs in each session should be synchronized with each other. This is accomplished by comparing the CNAME fields in the RTCP source description (SDES) packets. SSRCs with the same CNAME sent in any of multiple RTP sessions can be synchronized.
The actual RTCP mechanism for inter-media synchronization is based on the idea that each RTP stream provides a position on the media
specific time line (measured in RTP timestamp ticks) and a common reference time line. The common reference time line is expressed in RTCP as a wall-clock time in the Network Time Protocol (NTP) format.
It is important to notice that the wall-clock time is not required to be synchronized between hosts, for example, by using NTP [RFC5905].
It can even have nothing at all to do with the actual time; for
example, the host system’s up-time can be used for this purpose. The important factor is that all media streams from a particular source that are being synchronized use the same reference clock to derive their relative RTP timestamp time scales. The type of reference clock and its timebase can be signaled using RTP Clock Source Signaling [RFC7273].
Figure 1 illustrates how if one receives RTCP Sender Report (SR) packet P1 for one RTP stream and RTCP SR packet P2 for the other RTP stream, then one can calculate the corresponding RTP timestamp values for any arbitrary point in time T. However, to be able to do that, it is also required to know the RTP timestamp rates for each RTP stream currently used in the sessions.
TS1 --+---+--->
| | P1 | | |
NTP ---+---+---T--->
| | P2 | | |
TS2 ---+---+---X-->
Figure 1: RTCP Synchronization
Assume that medium 1 uses an RTP timestamp clock rate of 16 kHz, and medium 2 uses a clock rate of 90 kHz. Then, TS1 and TS2 for point T can be calculated in the following way: TS1(T) = TS1(P1) + 16000 * (NTP(T)-NTP(P1)) and TS2(T) = TS2(P2) + 90000 * (NTP(T)-NTP(P2)).
This calculation is useful as it allows the implementation to
generate a common synchronization point for which all time values are provided (TS1(T), TS2(T) and T). So, when one wishes to calculate the NTP time that the timestamp value present in packet X corresponds to, one can do that in the following way: NTP(X) = NTP(T) + (TS2(X) - TS2(T))/90000.
Improved signaling for layered codecs and fast tune-in have been specified in "Rapid Synchronization for RTP Flows" [RFC6051].
Leap seconds are extra seconds added or seconds removed to keep our clocks in sync with the earth’s rotation. Adding or removing seconds can impact the reference clock as discussed in "RTP and Leap Seconds"
[RFC7164]; also, in cases where the RTP timestamp values are derived using the wall clock during the leap second event, errors can occur.
Implementations need to consider leap seconds and should consider the recommendations in [RFC7164].
3.4. Signaling Aspects
RTP payload formats are used in the context of application signaling protocols such as SIP [RFC3261] using the Session Description
Protocol (SDP) [RFC4566] with Offer/Answer [RFC3264], RTSP [RFC7826], or the Session Announcement Protocol [RFC2974]. These examples all use out-of-band signaling to indicate which type of RTP streams are desired to be used in the session and how they are configured. To be able to declare or negotiate the media format and RTP payload
packetization, the payload format must be given an identifier. In addition to the identifier, many payload formats also have the need to signal further configuration information out-of-band for the RTP payloads prior to the media transport session.
The above examples of session-establishing protocols all use SDP, but other session description formats may be used. For example, there was discussion of a new XML-based session description format within the IETF (SDP-NG). In the end, the proposal did not get beyond draft protocol specification because of the enormous installed base of SDP implementations. However, to avoid locking the usage of RTP to SDP based out-of-band signaling, the payload formats are identified using a separate definition format for the identifier and associated
parameters. That format is the media type.
3.4.1. Media Types
Media types [RFC6838] are identifiers originally created for identifying media formats included in email. In this usage, they were known as MIME types, where the expansion of the MIME acronym includes the word "mail". The term "media type" was introduced to reflect a broader usage, which includes HTTP [RFC7231], Message Session Relay Protocol (MSRP) [RFC4975], and many other protocols to identify arbitrary content carried within the protocols. Media types also provide a media hierarchy that fits RTP payload formats well.
Media type names are of two parts and consist of content type and sub-type separated with a slash, e.g., ’audio/PCMA’ or ’video/
h263-2000’. It is important to choose the correct content-type when creating the media type identifying an RTP payload format. However, in most cases, there is little doubt what content type the format belongs to. Guidelines for choosing the correct media type and registration rules for media type names are provided in "Media Type Specifications and Registration Procedures" [RFC6838]. The
additional rules for media types for RTP payload formats are provided in "Media Type Registration of RTP Payload Formats" [RFC4855].
Registration of the RTP payload name is something that is required to avoid name collision in the future. Note that "x-" names are not suitable for any documented format as they have the same problem with name collision and can’t be registered. The list of already-
registered media types can be found at
<https://www.iana.org/assignments/media-types/media-types.xhtml>.
Media types are allowed any number of parameters, which may be
required or optional for that media type. They are always specified on the form "name=value". There exist no restrictions on how the value is defined from the media type’s perspective, except that parameters must have a value. However, the usage of media types in
SDP, etc., has resulted in the following restrictions that need to be followed to make media types usable for RTP-identifying payload
formats:
1. Arbitrary binary content in the parameters is allowed, but it needs to be encoded so that it can be placed within text-based protocols. Base64 [RFC4648] is recommended, but for shorter content Base16 [RFC4648] may be more appropriate as it is simpler to interpret for humans. This needs to be explicitly stated when defining a media type parameter with binary values.
2. The end of the value needs to be easily found when parsing a message. Thus, parameter values that are continuous and not interrupted by common text separators, such as space and
semicolon characters, are recommended. If that is not possible, some type of escaping should be used. Usage of quote (") is recommended; do not forget to provide a method of encoding any character used for quoting inside the quoted element.
3. A common representation form for the media type and its
parameters is on a single line. In that case, the media type is followed by a semicolon-separated list of the parameter value pairs, e.g.:
audio/amr octet-align=0; mode-set=0,2,5,7; mode-change-period=2 3.4.2. Mapping to SDP
Since SDP [RFC4566] is so commonly used as an out-of-band signaling protocol, a mapping of the media type into SDP exists. The details on how to map the media type and its parameters into SDP are
described in [RFC4855]. However, this is not sufficient to explain how certain parameters must be interpreted, for example, in the context of Offer/Answer negotiation [RFC3264].
3.4.2.1. The Offer/Answer Model
The Offer/Answer (O/A) model allows SIP to negotiate which media formats and payload formats are to be used in a session and how they are to be configured. However, O/A does not define a default
behavior; instead, it points out the need to define how parameters behave. To make things even more complex, the direction of media within a session has an impact on these rules, so that some cases may require separate descriptions for RTP streams that are send-only, receive-only, or both sent and received as identified by the SDP attributes a=sendonly, a=recvonly, and a=sendrecv. In addition, the usage of multicast adds further limitations as the same RTP stream is
delivered to all participants. If those multicast-imposed
restrictions are too limiting for unicast, then separate rules for unicast and multicast will be required.
The simplest and most common O/A interpretation is that a parameter is defined to be declarative; i.e., the SDP Offer/Answer sending agent can declare a value and that has no direct impact on the other agent’s values. This declared value applies to all media that are going to be sent to the declaring entity. For example, most video codecs have a level parameter that tells the other participants the highest complexity the video decoder supports. The level parameter can be declared independently by two participants in a unicast
session as it will be the media sender’s responsibility to transmit a video stream that fulfills the limitation the other side has
declared. However, in multicast, it will be necessary to send a stream that follows the limitation of the weakest receiver, i.e., the one that supports the lowest level. To simplify the negotiation in these cases, it is common to require any answerer to a multicast session to take a yes or no approach to parameters.
A "negotiated" parameter is a different case, for which both sides need to agree on its value. Such a parameter requires the answerer to either accept it as it is offered or remove the payload type the parameter belonged to from its answer. The removal of the payload type from the answer indicates to the offerer the lack of support for the parameter values presented. An unfortunate implication of the need to use complete payload types to indicate each possible
configuration so as to maximize the chances of achieving
interoperability, is that the number of necessary payload types can quickly grow large. This is one reason to limit the total number of sets of capabilities that may be implemented.
The most problematic type of parameters are those that relate to the media the entity sends. They do not really fit the O/A model, but can be shoehorned in. Examples of such parameters can be found in the H.264 video codec’s payload format [RFC6184], where the name of all parameters with this property starts with "sprop-". The issue with these parameters is that they declare properties for a RTP
stream that the other party may not accept. The best one can make of the situation is to explain the assumption that the other party will accept the same parameter value for the media it will receive as the offerer of the session has proposed. If the answerer needs to change any declarative parameter relating to streams it will receive, then the offerer may be required to make a new offer to update the
parameter values for its outgoing RTP stream.
Another issue to consider is the send-only RTP streams in offers.
Parameters that relate to what the answering entity accepts to receive have no meaning other than to provide a template for the answer. It is worth pointing out in the specification that these really provide a set of parameter values that the sender recommends.
Note that send-only streams in answers will need to indicate the offerer’s parameters to ensure that the offerer can match the answer to the offer.
A further issue with Offer/Answer that complicates things is that the answerer is allowed to renumber the payload types between offer and answer. This is not recommended, but allowed for support of gateways to the ITU conferencing suite. This means that it must be possible to bind answers for payload types to the payload types in the offer even when the payload type number has been changed, and some of the proposed payload types have been removed. This binding must normally be done by matching the configurations originally offered against those in the answer. This may require specification in the payload format of which parameters that constitute a configuration, for example, as done in Section 8.2.2 of the H.264 RTP Payload format [RFC6184], which states: "The parameters identifying a media format configuration for H.264 are profile-level-id and packetization-mode".
3.4.2.2. Declarative Usage in RTSP and SAP
SAP (Session Announcement Protocol) [RFC2974] was experimentally used for announcing multicast sessions. Similar but better protocols are using SDP in a declarative style to configure multicast-based
applications. Independently of the usage of Source-Specific Multicast (SSM) [RFC3569] or Any-Source Multicast (ASM), the SDP provided by these configuration delivery protocols applies to all participants. All media that is sent to the session must follow the RTP stream definition as specified by the SDP. This enables everyone to receive the session if they support the configuration. Here, SDP provides a one-way channel with no possibility to affect the
configuration that the session creator has decided upon. Any RTP payload format that requires parameters for the send direction and that needs individual values per implementation or instance will fail in a SAP session for a multicast session allowing anyone to send.
Real-Time Streaming Protocol (RTSP) [RFC7826] allows the negotiation of transport parameters for RTP streams that are part of a streaming session between a server and client. RTSP has divided the transport parameters from the media configuration. SDP is commonly used for media configuration in RTSP and is sent to the client prior to session establishment, either through use of the DESCRIBE method or
by means of an out-of-band channel like HTTP, email, etc. The SDP is used to determine which RTP streams and what formats are being used prior to session establishment.
Thus, both SAP and RTSP use SDP to configure receivers and senders with a predetermined configuration for a RTP stream including the payload format and any of its parameters. All parameters are used in a declarative fashion. This can result in different treatment of parameters between Offer/Answer and declarative usage in RTSP and SAP. Any such difference will need to be spelled out by the payload format specification.
3.5. Transport Characteristics
The general channel characteristics that RTP flows experience are documented in Section 3 of "Guidelines for Writers of RTP Payload Format Specifications" [RFC2736]. The discussion below provides additional information.
3.5.1. Path MTU
At the time of writing, the most common IP Maximum Transmission Unit (MTU) in commonly deployed link layers is 1500 bytes (Ethernet data payload). However, there exist both links with smaller MTUs and links with much larger MTUs. An example for links with small MTU size is older generation cellular links. Certain parts of the Internet already support an IP MTU of 8000 bytes or more, but these are limited islands. The most likely places to find MTUs larger than 1500 bytes are within enterprise networks, university networks, data centers, storage networks, and over high capacity (10 Gbps or more) links. There is a slow, ongoing evolution towards larger MTU sizes.
However, at the same time, it has become common to use tunneling protocols, often multiple ones, whose overhead when added together can shrink the MTU significantly. Thus, there exists a need both to consider limited MTUs as well as enable support of larger MTUs. This should be considered in the design, especially in regard to features such as aggregation of independently decodable data units.
3.5.2. Different Queuing Algorithms
Routers and switches on the network path between an IP sender and a particular receiver can exhibit different behaviors affecting the end-to-end characteristics. One of the more important aspects of this is queuing behavior. Routers and switches have some amount of queuing to handle temporary bursts of data that designated to leave the switch or router on the same egress link. A queue, when not
The implementation of the queuing affects the delay and also how congestion signals (Explicit Congestion Notification (ECN) [RFC6679]
or packet drops) are provided to the flow. The other aspects are if the flow shares the queue with other flows and how the implementation affects the flow interaction. This becomes important, for example, when real-time flows interact with long-lived TCP flows. TCP has a built-in behavior in its congestion control that strives to fill the buffer; thus, all flows sharing the buffer experienced the delay build up.
A common, but quite poor, queue-handling mechanism is tail-drop, i.e., only drop packets when the incoming packet doesn’t fit in the queue. If a bad queuing algorithm is combined with too much queue space, the queuing time can grow to be very significant and can even become multiple seconds. This is called "bufferbloat" [BLOAT].
Active Queue Management (AQM) is a term covering mechanisms that try to do something smarter by actively managing the queue, for example, sending congestion signals earlier by dropping packets earlier in the queue. The behavior also affects the flow interactions. For
example, Random Early Detection (RED) [RED] selects which packet(s) to drop randomly. This gives flows that have more packets in the queue a higher probability to experience the packet loss (congestion signal). There is ongoing work in the IETF WG AQM to find suitable mechanisms to recommend for implementation and reduce the use of tail-drop.
3.5.3. Quality of Service
Using best-effort Internet has no guarantees for the path’s
properties. QoS mechanisms are intended to provide the possibility to bound the path properties. Where Diffserv [RFC2475] markings affect the queuing and forwarding behaviors of routers, the mechanism provides only statistical guarantees and care in how much marked packets of different types that are entering the network. Flow-based QoS, like IntServ [RFC1633], has the potential for stricter
guarantees as the properties are agreed on by each hop on the path, at the cost of per-flow state in the network.
4. Standardization Process for an RTP Payload Format
This section discusses the recommended process to produce an RTP payload format in the described venues. This is to document the best current practice on how to get a well-designed and specified payload format as quickly as possible. For specifications that are defined by standards bodies other than the IETF, the primary milestone is the registration of the media type for the RTP payload format. For
proprietary media formats, the primary goal depends on whether
interoperability is desired at the RTP level. However, there is also the issue of ensuring best possible quality of any specification.
4.1. IETF
For all standardized media formats, it is recommended that the payload format be specified in the IETF. The main reason is to
provide an openly available RTP payload format specification that has been reviewed by people experienced with RTP payload formats. At the time of writing, this work is done in the PAYLOAD Working Group (WG), but that may change in the future.
4.1.1. Steps from Idea to Publication
There are a number of steps that an RTP payload format should go through from the initial idea until it is published. This also documents the process that the PAYLOAD WG applies when working with RTP payload formats.
Idea: Determine the need for an RTP payload format as an IETF specification.
Initial effort: Using this document as a guideline, one should be able to get started on the work. If one’s media codec doesn’t fit any of the common design patterns or one has problems
understanding what the most suitable way forward is, then one should contact the PAYLOAD WG and/or the WG Chairs. The goal of this stage is to have an initial individual draft. This draft needs to focus on the introductory parts that describe the real- time media format and the basic idea on how to packetize it. Not all the details are required to be filled in. However, the
security chapter is not something that one should skip, even initially. From the start, it is important to consider any
serious security risks that need to be solved. The first step is completed when one has a draft that is sufficiently detailed for a first review by the WG. The less confident one is of the
solution, the less work should be spent on details; instead, concentrate on the codec properties and what is required to make the packetization work.
Submission of the first version: When one has performed the above, one submits the draft as an individual draft
(https://datatracker.ietf.org/submit/). This can be done at any time, except for a period prior to an IETF meeting (see important dates related to the next IETF meeting for draft submission cutoff
the draft announcement list
(https://www.ietf.org/mailman/listinfo/I-D-Announce), forward it to the PAYLOAD WG (https://www.ietf.org/mailman/listinfo/payload) and request that it be reviewed. In the email, outline any issues the authors currently have with the design.
Iterative improvements: Taking the feedback received into account, one updates the draft and tries resolve issues. New revisions of the draft can be submitted at any time (again except for a short period before meetings). It is recommended to submit a new
version whenever one has made major updates or has new issues that are easiest to discuss in the context of a new draft version.
Becoming a WG document: Given that the definition of RTP payload formats is part of the PAYLOAD WG’s charter, RTP payload formats that are going to be published as Standards Track RFCs need to become WG documents. Becoming a WG document means that the WG Chairs or an appointed document shepherd are responsible for administrative handling, for example, issuing publication requests. However, be aware that making a document into a WG document changes the formal ownership and responsibility from the individual authors to the WG. The initial authors normally
continue being the document editors, unless unusual circumstances occur. The PAYLOAD WG accepts new RTP payload formats based on their suitability and document maturity. The document maturity is a requirement to ensure that there are dedicated document editors and that there exists a good solution.
Iterative improvements: The updates and review cycles continue until the draft has reached the level of maturity suitable for
publication. The authors are responsible for judging when the document is ready for the next step, most likely WG Last Call, but they can ask the WG chairs or Shepherd.
WG Last Call: A WG Last Call of at least two weeks is always performed for payload formats in the PAYLOAD WG (see Section 7.4 of [RFC2418]). The authors request WG Last Call for a draft when they think it is mature enough for publication. The WG Chairs or shepherd perform a review to check if they agree with the authors’
assessment. If the WG Chairs or shepherd agree on the maturity, the WG Last Call is announced on the WG mailing list. If there are issues raised, these need to be addressed with an updated draft version. For any more substantial changes to the draft, a new WG Last Call is announced for the updated version. Minor changes, like editorial fixes, can be progressed without an additional WG Last Call.
Publication requested: For WG documents, the WG Chairs or shepherd request publication of the draft after it has passed WG Last Call.
After this, the approval and publication process described in BCP 9 [BCP9] is performed. The status after the publication has been requested can be tracked using the IETF Datatracker [TRACKER].
Documents do not expire as they normally do after publication has been requested, so authors do not have to issue keep-alive
updates. In addition, any submission of document updates requires the approval of WG Chair(s). The authors are commonly asked to address comments or issues raised by the IESG. The authors also do one last review of the document immediately prior to its publication as an RFC to ensure that no errors or formatting problems have been introduced during the publication process.
4.1.2. WG Meetings
WG meetings are for discussing issues, not presentations. This means that most RTP payload formats should never need to be discussed in a WG meeting. RTP payload formats that would be discussed are either those with controversial issues that failed to be resolved on the mailing list or those including new design concepts worth a general discussion.
There exists no requirement to present or discuss a draft at a WG meeting before it becomes published as an RFC. Thus, even authors who lack the possibility to go to WG meetings should be able to successfully specify an RTP payload format in the IETF. WG meetings may become necessary only if the draft gets stuck in a serious debate that cannot easily be resolved.
4.1.3. Draft Naming
To simplify the work of the PAYLOAD WG Chairs and WG members, a specific Internet-Draft file-naming convention shall be used for RTP payload formats. Individual submissions shall be named using the template: draft-<lead author family name>-payload-rtp-<descriptive name>-<version>. The WG documents shall be named according to this template: draft-ietf-payload-rtp-<descriptive name>-<version>. The inclusion of "payload" in the draft file name ensures that the search for "payload-" will find all PAYLOAD-related drafts. Inclusion of "rtp" tells us that it is an RTP payload format draft. The
descriptive name should be as short as possible while still
describing what the payload format is for. It is recommended to use the media format or codec abbreviation. Please note that the version must start at 00 and is increased by one for each submission to the IETF secretary of the draft. No version numbers may be skipped. For
4.1.4. Writing Style
When writing an Internet-Draft for an RTP payload format, one should observe some few considerations (that may be somewhat divergent from the style of other IETF documents and/or the media coding spec’s author group may use):
Include Motivations: In the IETF, it is common to include the motivation for why a particular design or technical path was chosen. These are not long statements: a sentence here and there explaining why suffice.
Use the Defined Terminology: There exists defined terminology both in RTP and in the media codec specification for which the RTP payload format is designed. A payload format specification needs to use both to make clear the relation of features and their functions. It is unwise to introduce or, worse, use without introduction, terminology that appears to be more accessible to average readers but may miss certain nuances that the defined terms imply. An RTP payload format author can assume the reader to be reasonably familiar with the terminology in the media coding specification.
Keeping It Simple: The IETF has a history of specifications that are focused on their main usage. Historically, some RTP payload
formats have a lot of modes and features, while the actual deployments have only included the most basic features that had very clear requirements. Time and effort can be saved by focusing on only the most important use cases and keeping the solution simple. An extension mechanism should be provided to enable backward-compatible extensions, if that is an organic fit.
Normative Requirements: When writing specifications, there is commonly a need to make it clear when something is normative and at what level. In the IETF, the most common method is to use "Key words for use in RFCs to Indicate Requirement Levels" [RFC2119], which defines the meaning of "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL".
4.1.5. How to Speed Up the Process
There a number of ways to lose a lot of time in the above process.
This section discusses what to do and what to avoid.
o Do not update the draft only for the meeting deadline. An update to each meeting automatically limits the draft to three updates per year. Instead, ignore the meeting schedule and publish new versions as soon as possible.
o Try to avoid requesting reviews when people are busy, like the few weeks before a meeting. It is actually more likely that people have time for them directly after a meeting.
o Perform draft updates quickly. A common mistake is that the authors let the draft slip. By performing updates to the draft text directly after getting resolution on an issue, things speed up. This minimizes the delay that the author has direct control over. The time taken for reviews, responses from Area Directors and WG Chairs, etc., can be much harder to speed up.
o Do not fail to take human nature into account. It happens that people forget or need to be reminded about tasks. Send a kind reminder to the people you are waiting for if things take longer than expected. Ask people to estimate when they expect to fulfill the requested task.
o Ensure there is enough review. It is common that documents take a long time and many iterations because not enough review is
performed in each iteration. To improve the amount of review you get on your own document, trade review time with other document authors. Make a deal with some other document author that you will review their draft if they review yours. Even inexperienced reviewers can help with language, editorial, or clarity issues.
Also, try approaching the more experienced people in the WG and getting them to commit to a review. The WG Chairs cannot, even if desirable, be expected to review all versions. Due to workload, the Chairs may need to concentrate on key points in a draft evolution like checking on initial submissions, a draft’s
readiness to become a WG document, or its readiness for WG Last Call.
4.2. Other Standards Bodies
Other standards bodies may define RTP payloads in their own
specifications. When they do this, they are strongly recommended to
